[asterisk-users] Unable to make outgoing calls with Internode
Da Rock
asterisk-users at herveybayaustralia.com.au
Wed Feb 9 23:16:19 CST 2011
On 02/10/11 14:00, Dovid Bender wrote:
> Hi,
>
> Under sip-out why do you have secret, fromdomain and NAT commented out ?
>
> Also it seems like Asterisk is re-transmitting which means it seems
> like it is not getting any response from your ISP. It could be a
> firewall issue, it could be your ISP. If your ISP refuses to work with
> you you may want to go with an ISP that will help.
>
> Regards,
>
> Dovid
Thanks Dovid. I've actually tried others and they're surprisingly worse.
I haven't stopped going through settings and logs, etc; but I just
looked at them again fresh. I noticed fragmented packets which the
firewall was dropping. I tweaked it and it came good- finally!
I suspected it was a the case, but I just could never find where. What
has me wondering now is, why is asterisk fragmenting packets? Any other
packets- from the same machine even- are ok. Is it a bug? Have I
seriously been tearing my hair out because of a bug?
The settings are commented out because I have tried them, discarded
them, tried them again differently.... I left them in here so you could
see I might have tried them at one stage.
One more question- what is the issue with having a single peer reference
in sip.conf? Why does it say have one incoming and one outgoing? Is it a
problem just having the one?
Thanks again Dovid, you saved my sanity. Having no one to confirm or
deny a supposition can be a pain at times. Your confirmation helped
immensely.
Cheers
>
> ----- Original Message ----- From: "Da Rock"
> <asterisk-users at herveybayaustralia.com.au>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Thursday, February 10, 2011 05:08
> Subject: [asterisk-users] Unable to make outgoing calls with Internode
>
>
>> Surely there must be someone here who can help me with this problem.
>>
>> I have spent weeks trying to get this damned service to work with no
>> luck. I have incoming calls working, but no outgoing. If get outgoing
>> working then incoming don't work.
>>
>> I have sent this problem to this list a couple of times with little
>> or no response, and I _really_ need some help to sort it out.
>>
>> I have an asterisk 1.8 server running on FreeBSD 8.1, and another
>> FreeBSD 8.1 running as a firewall/gateway with PF.
>>
>> I have a nodephone service with Internode (who have been absolutely
>> useless in helping me- they point blank refuse, or they say to open
>> everything right up to their server; which didn't wok anyway btw).
>>
>> I have been running endless tests on settings changes, tcpdumps on
>> both the firewall and asterisk, and hours poring over SIP rfc's. I've
>> only managed to get a headache...
>>
>> I have tried following best practices, worst practices, and still
>> nothing works.
>>
>> My sip.conf looks like this:
>>
>> [general]
>> context = default
>> bindport = 5060
>> bindaddr = 0.0.0.0
>> srvlookup = yes
>> allow = all
>> ;allow = t140red
>> textsupport = yes
>> videosupport = yes
>> ;allow = h263
>> maxcallbitrate = 384
>> register => sip-in?<phone
>> number>:<secret>@sip.internode.on.net/<phone number>
>> externip = <my static ip>
>> localnet = <my local subnet>
>> canreinvite = no
>> hasvoicemail = no
>> qualify = yes
>> nat = no
>> ;rtptimeout = 120
>> rtpkeepalive = 5
>> ;ignoresdpversion = yes
>> ;directmediapermit = <my local subnet>
>>
>> [sip-in]
>> type = peer
>> host = sip.internode.on.net
>> context = internode-incoming
>> ;externip = <my static ip>
>> ;domain = internode.on.net,internode-incoming
>> ;fromdomain = sip.internode.on.net
>> ;fromuser = <phone number>
>> ;username = <phone number>
>> ;secret = <secret>
>> ;auth = <phone number>:<secret>@BroadWorks
>> ;insecure = invite,port
>> ;register => <phone number>:<secret>@sip.internode.on.net
>> ;nat = never
>> qualify = yes
>> canreinvite = no
>> ;expire = 240
>>
>> [sip-out]
>> type = peer
>> host = sip.internode.on.net
>> context = internode-outgoing
>> externip = <my static ip>
>> ;username = <phone number>
>> fromuser = <phone number>
>> ;fromdomain = internode.on.net
>> ;secret = <secret>
>> ;qualify = yes
>> canreinvite = no
>> ;auth = <phone number>:<secret>@BroadWorks
>> ;nat = never
>> ;pedantic = yes
>> ;insecure = invite,port
>> ;ignoresdpversion = yes
>> ;compactheaders = yes
>>
>> As you can see I've tried lots of settings. It registers and peers
>> with the provider, but no outgoing. The provider can call me though.
>>
>> In extensions.conf:
>>
>> [internode-outgoing]
>> exten => _X.,1,Dial(SIP/${EXTEN}@sip-out)
>> exten => _X.,n,Answer(2)
>> exten => _X.,n,Playback(ss-noservice)
>>
>> With debugging enabled, verbose 9, debug 9:
>>
>> SIP Debugging enabled
>>
>> <--- SIP read from UDP:<my ata ip>:5060 --->
>> INVITE sip:0871271201@<asterisk server> SIP/2.0
>> Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-78cdde11;rport
>> From: <my ata cid> <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>
>> Call-ID: e2895c9d-55b90b64@<my ata ip>
>> CSeq: 101 INVITE
>> Max-Forwards: 70
>> Contact: <my ata cid> <sip:<my ata username>@<my ata ip>:5060>
>> Expires: 240
>> User-Agent: Linksys/PAP2T-3.1.15(LS)
>> Content-Length: 446
>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>> Supported: x-sipura
>> Content-Type: application/sdp
>>
>> v=0
>> o=- 5330142 5330142 IN IP4 <my ata ip>
>> s=-
>> c=IN IP4 <my ata ip>
>> t=0 0
>> m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:2 G726-32/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729a/8000
>> a=rtpmap:96 G726-40/8000
>> a=rtpmap:97 G726-24/8000
>> a=rtpmap:98 G726-16/8000
>> a=rtpmap:100 NSE/8000
>> a=fmtp:100 192-193
>> a=rtpmap:101 telephone-event/--- (14 headers 20 lines) ---
>> Sending to <my ata ip>:5060 (no NAT)
>> Using INVITE request as basis request - e2895c9d-55b90b64@<my ata ip>
>> Found peer '<my ata username>' for '<my ata username>' from <my ata
>> ip>:5060
>>
>> <--- Reliably Transmitting (no NAT) to <my ata ip>:5060 --->
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP <my ata
>> ip>:5060;branch=z9hG4bK-78cdde11;received=<my ata ip>;rport=5060
>> From: Skinner's Home <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>;tag=as6957dfb9
>> Call-ID: e2895c9d-55b90b64 at 192.168.0.196
>> CSeq: 101 INVITE
>> Server: Asterisk PBX 1.8.1.1
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
>> nonce="12eb6973"
>> Content-Length: 0
>>
>>
>> <------------>
>> Scheduling destruction of SIP dialog 'e2895c9d-55b90b64@<my ata ip>'
>> in 6400 ms (Method: INVITE)
>>
>> <--- SIP read from UDP:<my ata ip>:5060 --->
>> ACK sip:0871271201@<asterisk server> SIP/2.0
>> Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-78cdde11;rport
>> From: <my ata cid> <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>;tag=as6957dfb9
>> Call-ID: e2895c9d-55b90b64@<my ata ip>
>> CSeq: 101 ACK
>> Max-Forwards: 70
>> Contact: <my ata cid> <sip:<my ata username>@<my ata ip>:5060>
>> User-Agent: Linksys/PAP2T-3.1.15(LS)
>> Content-Length: 0
>>
>> <------------->
>> --- (10 headers 0 lines) ---
>>
>> <--- SIP read from UDP:<my ata ip>:5060 --->
>> INVITE sip:0871271201@<asterisk server> SIP/2.0
>> Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-6035d0b9;rport
>> From: <my ata cid> <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>
>> Call-ID: e2895c9d-55b90b64@<my ata ip>
>> CSeq: 102 INVITE
>> Max-Forwards: 70
>> Authorization: Digest username="<my ata
>> username>",realm="asterisk",nonce="12eb6973",uri="sip:0871271201@<asterisk
>> server>",algorithm=MD5,response="aa3d9a1719fee78526adb69c56472ceb"
>> Contact: <my ata cid> <sip:<my ata username>@<my ata ip>:5060>
>> Expires: 240
>> User-Agent: Linksys/PAP2T-3.1.15(LS)
>> Content-Length: 446
>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>> Supported: x-sipura
>> Content-Type: application/sdp
>>
>> v=0
>> o=- 5330142 5330142 IN IP4 <my ata ip>
>> s=-
>> c=IN IP4 <my ata ip>
>> t=0 0
>> m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:2 G726-32/8000
>> a=rtpmap:4 G723/8000
>> --- (15 headers 20 lines) ---
>> Sending to <my ata ip>:5060 (no NAT)
>> Using INVITE request as basis request - e2895c9d-55b90b64@<my ata ip>
>> Found peer '<my ata username>' for '<my ata username>' from <my ata
>> ip>:5060
>> Found RTP audio format 0
>> Found RTP audio format 2
>> Found RTP audio format 4
>> Found RTP audio format 8
>> Found RTP audio format 18
>> Found RTP audio format 96
>> Found RTP audio format 97
>> Found RTP audio format 98
>> Found RTP audio format 100
>> Found RTP audio format 101
>> Found audio description format PCMU for ID 0
>> Found audio description format G726-32 for ID 2
>> Found audio description format G723 for ID 4
>> Found audio description format PCMA for ID 8
>> Found audio description format G729a for ID 18
>> Found audio description format G726-40 for ID 96
>> Found audio description format G726-24 for ID 97
>> Found audio description format G726-16 for ID 98
>> Found audio description format NSE for ID 100
>> Found audio description format telephone-event for ID 101
>> Capabilities: us - 0x80030c7fffff
>> (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719),
>> peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0
>> (nothing)/text=0x0 (nothing), combined - 0x100d0d
>> (g723|ulaw|alaw|g726|g729|ilbc|h263p)
>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer -
>> 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
>> Peer audio RTP is at port <my ata ip>:16436
>> Peer doesn't provide video
>> Peer doesn't provide T.140
>> Looking for 0871271201 in users (domain <asterisk server>)
>> list_route: hop: <sip:<my ata username>@<my ata ip>:5060>
>>
>> <--- Transmitting (no NAT) to <my ata ip>:5060 --->
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP <my ata
>> ip>:5060;branch=z9hG4bK-6035d0b9;received=<my ata ip>;rport=5060
>> From: <my ata cid> <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>
>> Call-ID: e2895c9d-55b90b64@<my ata ip>
>> CSeq: 102 INVITE
>> Server: Asterisk PBX 1.8.1.1
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Contact: <sip:0871271201@<asterisk server>:5060>
>> Content-Length: 0
>>
>>
>> <------------>
>> -- Executing [0871271201 at users:1] Goto("SIP/<my ata
>> username>-0000015c", "internode-outgoing,0871271201,1") in new stack
>> -- Goto (internode-outgoing,0871271201,1)
>> -- Executing [0871271201 at internode-outgoing:1] Dial("SIP/<my ata
>> username>-0000015c", "SIP/0871271201 at sip-out") in new stack
>> We think we can do text
>> And we have a text rtp object
>> Audio is at 5060
>> Video is at <my static ip>:5060
>> Lets set up the text sdp
>> Text is at <my static ip>:5060
>> Adding codec 0x4 (ulaw) to SDP
>> Adding codec 0x2 (gsm) to SDP
>> Adding codec 0x8 (alaw) to SDP
>> Adding codec 0x10 (g726aal2) to SDP
>> Adding codec 0x20 (adpcm) to SDP
>> Adding codec 0x40 (slin) to SDP
>> Adding codec 0x80 (lpc10) to SDP
>> Adding codec 0x200 (speex) to SDP
>> Adding codec 0x400 (ilbc) to SDP
>> Adding codec 0x800 (g726) to SDP
>> Adding codec 0x1000 (g722) to SDP
>> Adding codec 0x8000 (slin16) to SDP
>> Adding video codec 0x100000 (h263p) to SDP
>> Adding text codec 0x4000000 (red) to SDP
>> Adding text codec 0x8000000 (t140) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>> Reliably Transmitting (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK0c95373b
>> Max-Forwards: 70
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:<phone number>@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Date: Thu, 10 Feb 2011 02:04:14 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 738
>>
>> v=0
>> o=root 51098296 51098296 IN IP4 <my static ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <my static ip>
>> b=CT:384
>> t=0 0
>> m=audio 19850 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:112 AAL2-G726-32/8000
>> a=rtpmap:5 DVI4/8000
>> a=rtpmap:10 L16/8000
>> a=rtpmap:7 LPC/8000
>> a=rtpmap:110 speex/ -- Called 0871271201 at sip-out
>>
>> <--- SIP read from UDP:203.2.134.1:5060 --->
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK0c95373b
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 102 INVITE
>>
>> <------------->
>> --- (6 headers 0 lines) ---
>>
>> <--- SIP read from UDP:203.2.134.1:5060 --->
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK0c95373b
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>;tag=232999791-1297303507574
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 102 INVITE
>> WWW-Authenticate: DIGEST
>> qop="auth",nonce="BroadWorksXgjz10gvqTmcdnmtBW",realm="BroadWorks",algorithm=MD5
>>
>> Content-Length: 0
>>
>> <------------->
>> --- (8 headers 0 lines) ---
>> Transmitting (no NAT) to 203.2.134.1:5060:
>> ACK sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK0c95373b
>> Max-Forwards: 70
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>;tag=232999791-1297303507574
>> Contact: <sip:<phone number>@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 102 ACK
>> User-Agent: Asterisk PBX 1.8.1.1
>> Content-Length: 0
>>
>>
>> ---
>> We think we can do text
>> And we have a text rtp object
>> Audio is at 5060
>> Video is at <my static ip>:5060
>> Lets set up the text sdp
>> Text is at <my static ip>:5060
>> Adding codec 0x4 (ulaw) to SDP
>> Adding codec 0x2 (gsm) to SDP
>> Adding codec 0x8 (alaw) to SDP
>> Adding codec 0x10 (g726aal2) to SDP
>> Adding codec 0x20 (adpcm) to SDP
>> Adding codec 0x40 (slin) to SDP
>> Adding codec 0x80 (lpc10) to SDP
>> Adding codec 0x200 (speex) to SDP
>> Adding codec 0x400 (ilbc) to SDP
>> Adding codec 0x800 (g726) to SDP
>> Adding codec 0x1000 (g722) to SDP
>> Adding codec 0x8000 (slin16) to SDP
>> Adding video codec 0x100000 (h263p) to SDP
>> Adding text codec 0x4000000 (red) to SDP
>> Adding text codec 0x8000000 (t140) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>> Reliably Transmitting (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
>> Max-Forwards: 70
>> From: "<my ata username>" <sip:<phone number>@<my static
>> ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:<phone number>@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 103 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Authorization: Digest username="<phone number>", realm="BroadWorks",
>> algorithm=MD5, uri="sip:0871271201 at sip.internode.on.net",
>> nonce="BroadWorksXgjz10gvqTmcdnmtBW",
>> response="829397204c8dd43c8fc3435baa253075", qop=auth,
>> cnonce="03a77924", nc=00000001
>> Date: Thu, 10 Feb 2011 02:04:14 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 738
>>
>> v=0
>> o=root 51098296 51098297 IN IP4 <my static ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <my static ip>
>> b=CT:384
>> Retransmitting #1 (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
>> Max-Forwards: 70
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:0731292848@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 103 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Authorization: Digest username="<phone number>", realm="BroadWorks",
>> algorithm=MD5, uri="sip:0871271201 at sip.internode.on.net",
>> nonce="BroadWorksXgjz10gvqTmcdnmtBW",
>> response="829397204c8dd43c8fc3435baa253075", qop=auth,
>> cnonce="03a77924", nc=00000001
>> Date: Thu, 10 Feb 2011 02:04:14 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 738
>>
>> v=0
>> o=root 51098296 51098297 IN IP4 <my static ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <my static ip>
>> b=CT:384
>> t=0 0Retransmitting #2 (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
>> Max-Forwards: 70
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:<phone number>@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 103 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Authorization: Digest username="<phone number>", realm="BroadWorks",
>> algorithm=MD5, uri="sip:0871271201 at sip.internode.on.net",
>> nonce="BroadWorksXgjz10gvqTmcdnmtBW",
>> response="829397204c8dd43c8fc3435baa253075", qop=auth,
>> cnonce="03a77924", nc=00000001
>> Date: Thu, 10 Feb 2011 02:04:14 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 738
>>
>> v=0
>> o=root 51098296 51098297 IN IP4 <my static ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <my static ip>
>> b=CT:384
>> t=0 0Retransmitting #3 (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
>> Max-Forwards: 70
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:<phone number>@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 103 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Authorization: Digest username="<phone number>", realm="BroadWorks",
>> algorithm=MD5, uri="sip:0871271201 at sip.internode.on.net",
>> nonce="BroadWorksXgjz10gvqTmcdnmtBW",
>> response="829397204c8dd43c8fc3435baa253075", qop=auth,
>> cnonce="03a77924", nc=00000001
>> Date: Thu, 10 Feb 2011 02:04:14 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 738
>>
>> v=0
>> o=root 51098296 51098297 IN IP4 <my static ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <my static ip>
>> b=CT:384
>> t=0 0Retransmitting #4 (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
>> Max-Forwards: 70
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:<phone number>@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 103 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Authorization: Digest username="<phone number>", realm="BroadWorks",
>> algorithm=MD5, uri="sip:0871271201 at sip.internode.on.net",
>> nonce="BroadWorksXgjz10gvqTmcdnmtBW",
>> response="829397204c8dd43c8fc3435baa253075", qop=auth,
>> cnonce="03a77924", nc=00000001
>> Date: Thu, 10 Feb 2011 02:04:14 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 738
>>
>> v=0
>> o=root 51098296 51098297 IN IP4 <my static ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <my static ip>
>> b=CT:384
>> t=0 0Retransmitting #5 (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
>> Max-Forwards: 70
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:<phone number>@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 103 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Authorization: Digest username="<phone number>", realm="BroadWorks",
>> algorithm=MD5, uri="sip:0871271201 at sip.internode.on.net",
>> nonce="BroadWorksXgjz10gvqTmcdnmtBW",
>> response="829397204c8dd43c8fc3435baa253075", qop=auth,
>> cnonce="03a77924", nc=00000001
>> Date: Thu, 10 Feb 2011 02:04:14 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 738
>>
>> v=0
>> o=root 51098296 51098297 IN IP4 <my static ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <my static ip>
>> b=CT:384
>> t=0 0Retransmitting #6 (no NAT) to 203.2.134.1:5060:
>> INVITE sip:0871271201 at sip.internode.on.net SIP/2.0
>> Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
>> Max-Forwards: 70
>> From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
>> To: <sip:0871271201 at sip.internode.on.net>
>> Contact: <sip:<phone number>@<my static ip>:5060>
>> Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
>> CSeq: 103 INVITE
>> User-Agent: Asterisk PBX 1.8.1.1
>> Authorization: Digest username="<phone number>", realm="BroadWorks",
>> algorithm=MD5, uri="sip:0871271201 at sip.internode.on.net",
>> nonce="BroadWorksXgjz10gvqTmcdnmtBW",
>> response="829397204c8dd43c8fc3435baa253075", qop=auth,
>> cnonce="03a77924", nc=00000001
>> Date: Thu, 10 Feb 2011 02:04:14 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 738
>>
>> v=0
>> o=root 51098296 51098297 IN IP4 <my static ip>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <my static ip>
>> b=CT:384
>> t=0 0[Feb 10 12:04:20] WARNING[993]: chan_sip.c:3386 retrans_pkt:
>> Retransmission timeout reached on transmission
>> 784523d570058f2f64315e506a79ee0f@<my static ip>:5060 for seqno 103
>> (Critical Request) -- See doc/sip-retransmit.txt.
>> Packet timed out after 6400ms with no response
>> [Feb 10 12:04:20] WARNING[993]: chan_sip.c:3415 retrans_pkt: Hanging
>> up call 784523d570058f2f64315e506a79ee0f@<my static ip>:5060 - no
>> reply to our critical packet (see doc/sip-retransmit.txt).
>> == Everyone is busy/congested at this time (1:0/0/1)
>> -- Executing [0871271201 at internode-outgoing:2] Answer("SIP/<my
>> ata username>-0000015c", "2") in new stack
>> Audio is at 5060
>> Adding codec 0x1 (g723) to SDP
>> Adding codec 0x4 (ulaw) to SDP
>> Adding codec 0x8 (alaw) to SDP
>> Adding codec 0x100 (g729) to SDP
>> Adding codec 0x400 (ilbc) to SDP
>> Adding codec 0x800 (g726) to SDP
>> Adding video codec 0x100000 (h263p) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>>
>> <--- Reliably Transmitting (no NAT) to <my ata ip>:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP <my ata
>> ip>:5060;branch=z9hG4bK-6035d0b9;received=<my ata ip>;rport=5060
>> From: <my ata cid> <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>;tag=as6eeddb0e
>> Call-ID: e2895c9d-55b90b64@<my ata ip>
>> CSeq: 102 INVITE
>> Server: Asterisk PBX 1.8.1.1
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Contact: <sip:0871271201@<asterisk server>:5060>
>> Content-Type: application/sdp
>> Content-Length: 423
>>
>> v=0
>> o=root 1590196377 1590196377 IN IP4 <asterisk server>
>> s=Asterisk PBX 1.8.1.1
>> c=IN IP4 <asterisk server>
>> t=0 0
>> m=audio 10024 RTP/AVP 4 0 8 18 97 2 101
>> a=rtpmap:4 G723/8000
>> a=fmtp:4 annexa=no
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:97 iLBC/8000
>> a=fmtp:97 mode=30
>> a=rtpmap:2 G726-32/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fReally destroying SIP dialog '784523d570058f2f64315e506a79ee0f@<my
>> static ip>:5060' Method: INVITE
>>
>> <--- SIP read from UDP:<my ata ip>:5060 --->
>> ACK sip:0871271201@<asterisk server>:5060 SIP/2.0
>> Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-3b9bc888;rport
>> From: <my ata cid> <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>;tag=as6eeddb0e
>> Call-ID: e2895c9d-55b90b64@<my ata ip>
>> CSeq: 102 ACK
>> Max-Forwards: 70
>> Authorization: Digest username="<my ata
>> username>",realm="asterisk",nonce="12eb6973",uri="sip:0871271201@<asterisk
>> server>:5060",algorithm=MD5,response="c09a8c20894f257a63225f68d9ef54b7"
>> Contact: <my ata cid> <sip:<my ata username>@<my ata ip>:5060>
>> User-Agent: Linksys/PAP2T-3.1.15(LS)
>> Content-Length: 0
>>
>> <------------->
>> --- (11 headers 0 lines) ---
>> -- Executing [0871271201 at internode-outgoing:3] Playback("SIP/<my
>> ata username>-0000015c", "ss-noservice") in new stack
>> -- <SIP/<my ata usename>-0000015c> Playing 'ss-noservice.gsm'
>> (language 'en')
>>
>> <--- SIP read from UDP:<my ata ip>:5060 --->
>> BYE sip:0871271201@<asterisk server>:5060 SIP/2.0
>> Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-16d61cfc;rport
>> From: <my ata cid> <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>;tag=as6eeddb0e
>> Call-ID: e2895c9d-55b90b64@<my ata ip>
>> CSeq: 103 BYE
>> Max-Forwards: 70
>> Authorization: Digest username="<my ata
>> username>",realm="asterisk",nonce="12eb6973",uri="sip:0871271201@<asterisk
>> server>:5060",algorithm=MD5,response="bcad36f00cb422a4e856dec00d73e0d1"
>> User-Agent: Linksys/PAP2T-3.1.15(LS)
>> P-RTP-Stat:
>> PS=502,OS=40160,PR=226,OR=36160,PL=0,JI=0,LA=0,DU=5,EN=G711u,DE=G711u
>> Content-Length: 0
>>
>> <------------->
>> --- (11 headers 0 lines) ---
>> Sending to <my ata ip>:5060 (no NAT)
>> Scheduling destruction of SIP dialog 'e2895c9d-55b90b64@<my ata ip>'
>> in 6400 ms (Method: BYE)
>>
>> <--- Transmitting (no NAT) to <my ata ip>:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP <my ata
>> ip>:5060;branch=z9hG4bK-16d61cfc;received=<my ata ip>;rport=5060
>> From: <my ata cid> <sip:<my ata username>@<asterisk
>> server>>;tag=600053496208a4a8o1
>> To: <sip:0871271201@<asterisk server>>;tag=as6eeddb0e
>> Call-ID: e2895c9d-55b90b64@<my ata ip>
>> CSeq: 103 BYE
>> Server: Asterisk PBX 1.8.1.1
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Length: 0
>>
>>
>> <------------>
>> == Spawn extension (internode-outgoing, 0871271201, 3) exited
>> non-zero on 'SIP/<my ata username>-0000015c'
>>
>>
>> Names and identities have been masked to protect the innocent.
>>
>> My firewall is setup to binat between asterisk and the static ip, and
>> failing that my internal network (or dmz, which the asterisk is a
>> part of) is allowed outgoing traffic natted to the internet.
>>
>> I've opened up port 5060 and 10000:20000 to the outside world _only_
>> to the asterisk server, and the same outgoing.
>>
>> As near as I can tell asterisk simply can't auth with Internode for
>> some weird reason. The tcpdumps show 401 from internode, and later a
>> 408- sometimes. Or just a 408.
>>
>> The ata could connect, and tcpdumps show invite, 100, 401, then an
>> invite with auth, then 100, 180, and finally 200 and a conversation.
>>
>> According to internode they've changed the way it works by turning a
>> peer to peer connection into a client server model. But I don't think
>> asterisk is going to play that game.
>>
>> I _really_ need to see some light of day here. I am new to asterisk,
>> but I've been playing with firewalls for sometime now. A hint, a
>> clue, a solution- anything- would be helpful right about now.
>>
>> TIA
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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